[Freeswitch-users] Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN

Anthony Minessale anthony.minessale at gmail.com
Wed Sep 11 20:15:59 MSD 2013


Are you testing by connecting to internal or external IP?
Uncommenting the param is really the only thing you have to do.




On Wed, Sep 11, 2013 at 10:28 AM, James Mortensen <
james.mortensen at synclio.com> wrote:

> Hi Brian,
>
> I didn't stray too far from the default configuration.  I just uncommented
> the param and left the default port in place, 5066.  Is there a way to tell
> if the WS/HTTP server is running from within Freeswitch?  I didn't see a
> command for that.
>
> I also replaced the ext-sip-ip and ext-rtp-ip with my stun server and my
> server's public IP in internal.xml:
>
>     <param name="ext-rtp-ip" value="stun:stun.l.google.com:19302"/>
>     <param name="ext-sip-ip" value="MY_PUBLIC_IP" />
>
> The server starts up, but doesn't accept websocket connections.  What
> information can I get you so this is more helpful?
>
> Thank you!
> James
>
>
> James
>
> *"Every Call, Every Time!"*
>
> How did I do?
>
> [image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image:
> Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image:
> Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>
>
> Click on a face to provide feedback on my performance!<https://secure.teamhively.com/welcome/rate/735>
>
> Synclio Extension:*  110627*
> Phone: * 866-707-4590*
> Skype: *No!* *Call me on Synclio! :)*
>
>
> On Wed, Sep 11, 2013 at 6:13 AM, Brian West <brian at freeswitch.org> wrote:
>
>> What exactly did you put into the ws-binding param?
>>
>> /b
>>
>>
>> On Sep 10, 2013, at 8:48 PM, James Mortensen <james.mortensen at synclio.com>
>> wrote:
>>
>> > Hello,
>> >
>> > I'm working on a one-way audio bug that may or may not be either
>> Asterisk or Chrome WebRTC related. Details are here:
>> https://code.google.com/p/webrtc/issues/detail?id=2347, but basically
>> the problem is that I get about 1 in 10 calls returning one way audio where
>> audio flows from the PSTN to Chrome but not the other way.
>> >
>> > I believe the issue may be with Asterisk 11, based on Wireshark traces
>> and other troubleshooting, and if that's the case, I'm happy to try
>> Freeswitch as an alternative.
>> > I tried the demo here:
>> https://webrtc.freeswitch.org/webrtc/portal.html, and the 20 test calls
>> to the PSTN worked perfect with two way audio.
>> >
>> > So, now, onto the Freeswitch question:  Where is all the documentation
>> for configuring the Websocket server on Freeswitch 1.4b?  Where is the
>> documentation that explains how to get ICE enabled?  Why, when I enabled
>> the ws-bind port in internal.xml does the client fail to establish a WS
>> connection?
>> >
>> > Basically, I'd love to get my hands on the docs for setting this up,
>> but the Internet is largely silent on this matter?  Is it possible for the
>> Freeswitch team to share the configuration they used to get that
>> JsSIP/Freeswitch demo setup?
>> >
>> > If not the configuration, where would I find the docs for Freeswitch
>> 1.4b?
>> >
>> > Thank you!!
>> >
>> > James
>> >
>> > "Every Call, Every Time!"
>> >
>> > How did I do?
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130911/924a110b/attachment.html 


Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users mailing list